]> git.ipfire.org Git - thirdparty/asterisk.git/commitdiff
Fixes ringback tone on sip semi-attended transfer.
authorDavid Vossel <dvossel@digium.com>
Thu, 4 Nov 2010 21:39:51 +0000 (21:39 +0000)
committerDavid Vossel <dvossel@digium.com>
Thu, 4 Nov 2010 21:39:51 +0000 (21:39 +0000)
ABE-2168

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293924 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

index 81bc410ead6a73bcf1a40b2a2f507b93619b6660..6708a63ba029b03dbb02ed777bae917336425a0e 100644 (file)
@@ -21833,6 +21833,10 @@ static int local_attended_transfer(struct sip_pvt *transferer, struct sip_dual *
 
                ast_indicate(target.chan1, AST_CONTROL_UNHOLD);
 
+               if (current->chan2 && current->chan2->_state == AST_STATE_RING) {
+                       ast_indicate(target.chan1, AST_CONTROL_RINGING);
+               }
+
                if (target.chan2) {
                        ast_channel_queue_connected_line_update(target.chan1, &connected_to_transferee, NULL);
                        ast_channel_queue_connected_line_update(target.chan2, &connected_to_target, NULL);